Hi Geoff,
The best place for patches to Asterisk is the Asterisk bug tracker,
which can be found at http://bugs.digium.com .
*Please* don't forget to read the guidelines:
http://www.digium.com/bugguidelines.html and file a disclaimer.
Thanks for the patch! I'll look for it on the bug tracker :)
On Sat, 29 Jan 2005 16:44:16 -0600, Eric Wieling aka ManxPower
[EMAIL PROTECTED] wrote:
[EMAIL PROTECTED] wrote:
Folks,
Many thanks to Howard Lowndes who helped solve this problem; I ended
up using SetCallerID() instead of SetCIDName() as Howard suggested. Although
SetCIDName()
hi,
just got an strange crash, and don't know what could cause this type of crashs
- hardware failure
- memory
- cpu
?
i have 1xTE405P installed with 4xTA750. using fresh kernel 2.6.9 (no patch).
* version is latest CVS HEAD.
thanks
Program terminated with signal 11, Segmentation fault.
On Sun, 2005-01-30 at 12:31 +0330, Paradise Dove wrote:
hi,
just got an strange crash, and don't know what could cause this type of crashs
- hardware failure
- memory
- cpu
?
i have 1xTE405P installed with 4xTA750. using fresh kernel 2.6.9 (no patch).
* version is latest CVS HEAD.
In an attempt to eliminate audio echo I upgraded one side of a working
x-lite to x-lite
connection to eyebeam. No joy, and what was worse is the audio was even
worse
now - just noise. Ok, I upgraded the other side to eyebeam and same
thing. I'm
not even using video (will enable it in
this is what i've typed to get the crash info:
gdb /usr/sbin/asterisk --core=/core.3673
is it wrong?
On Sun, 30 Jan 2005 03:11:24 -0600, Steven Critchfield
[EMAIL PROTECTED] wrote:
On Sun, 2005-01-30 at 12:31 +0330, Paradise Dove wrote:
hi,
just got an strange crash, and don't know what
On Sun, 2005-01-30 at 12:46 +0330, Paradise Dove wrote:
this is what i've typed to get the crash info:
gdb /usr/sbin/asterisk --core=/core.3673
Not sure if that is wrong, but I also see from the gdb man page that you
should be able to start it by
gdb /usr/sbin/asterisk /core.3673
On Sun,
the same result!
On Sun, 30 Jan 2005 03:24:38 -0600, Steven Critchfield
[EMAIL PROTECTED] wrote:
On Sun, 2005-01-30 at 12:46 +0330, Paradise Dove wrote:
this is what i've typed to get the crash info:
gdb /usr/sbin/asterisk --core=/core.3673
Not sure if that is wrong, but I also see
Hi Paul,
I have the Cisco 7960 which has 6 lines. You can configure the other lines with
different extension numbers if you wish and then set your dialplan so that
calls coming in over different circuits, FWD, POTS etc come in different
buttons on your phone.
I'm sure there are lots of other uses
Anyone got any experiences of these with *, and also costings?
Thanks
John
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I have Asterisk 1.0.5-BRIstuffed-0.2.0-RC5 up and running. Everything
seems to be running fine but after some time asterisk just goes crazy
(even withouth any incoming or outgoing call activity perviously).
If I leave the box up for some time * goes haywire and the console is
flooded with this
Jeff,
does any email get sent out?
Mike
On Thu, 27 Jan 2005 18:42:25 -0500, Jeff R Glassman
[EMAIL PROTECTED] wrote:
I am running [EMAIL PROTECTED]
Voicemail works fine but does not email out the voicemail attachments. Any
suggestion?
---
Voicemail.conf
Hello,
What does :
X100P supports FXS Loopstart and Kewlstart
mean.
Thanks
Varun
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Hello,
Can the wildcard x100P be bought
of the shelf in stores - say in Singapore.
Thanks
Varun
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Citat Remco Barende [EMAIL PROTECTED]:
I have Asterisk 1.0.5-BRIstuffed-0.2.0-RC5 up and running. Everything
seems to be running fine but after some time asterisk just goes crazy
(even withouth any incoming or outgoing call activity perviously).
If I leave the box up for some time * goes
Martijn,
Please keep me posted on this issue, since I am very interested in resolution
of this problem. Thanks.
Vel
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk
Sent: Thursday, January 27, 2005 10:31 PM
To: Martijn van Oosterhout; Asterisk
You'll be better off buying the TDM400 from Digium.
I just ditched all my X100P clones because of huge problems with echoes.
Even worse, the interrupt problems with the clone cards caused my box to
crash under heavy I/O.
Want to buy the X100P clones I have? :)
On Sun, 30 Jan 2005 [EMAIL
You might try to run asterisk without having mpg123 on the system, see if that
gives the same problem.
I just removed the mp123 that was on the box. I also did a locate -i for
anyhthing mpg'ish on it, zip now! I also commenteded out res_musiconhold
in /etc/asterisk/modules will see how long the
Dont you need allow=ulaw before the register= line?
On Sat, 29 Jan 2005 13:13:28 -0700, Joseph [EMAIL PROTECTED] wrote:
When I try to call out to FWD over IAX2 I get:
Call rejected by 65.39.205.121: Unable to negotiate codec
I'm using asterisk-1.0.5 (the same settings works fine with *0.9)
Hi ALL;
Anybody knows how asterisk call forwarding to
another number works?
can you plz give me a sample config?
Regards
Mohammad
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Adam Hart wrote:
As always, I'm happy to announce a new version of Firefly.
Firefly 1.9.8 has more of what you want and less of what you don't
http://www.virbiage.com/firefly/download/firefly-thirdparty.exe
There's a few bug fixes - notably fixed the Reject button and sending of
audio before
I'm having a problem, I'm not sure if it has todo with the fact that my
phone is behind a NAT or not, but here it is..
My problem is when I call out, my asterisk system routes the call to my
SIP provider, whoever, as soon as the other party answers, asterisk
tries to make a native bridge
Anybody knows how asterisk call forwarding to another number works?
can you plz give me a sample config?
Depends on what you're trying to accomplish. Call forward when busy, call
forward on no answer, call forward when user press a phone button, call
forward after user enters a code (eg, *72),
Hi,
This might not be a very popular question, but I was just wondering if
anyone have ever tried to run Asterisk on a Windows computer using Microsoft
Virtual Server
(http://www.microsoft.com/windowsserversystem/virtualserver/default.mspx).
I am told that you can run Linux on a virtual server
Kewlstart is part of the spec for forward disconnect on loopstart lines in
North America(I am not familar with telco specs elsewhere in the world).
When the party at the other end of the call hangs up, the telco switch
removes talk battery from the circuit for a short period of time to signal
On Sat, 29 Jan 2005, Andrew Kohlsmith wrote:
On January 29, 2005 11:29 pm, Brian Dingman wrote:
This is driving me crazy. I have resorted to using the m option in the
Dial command just so folks don't hang up. I can't believe nobody else
is having this issue.
Simple test: try it with
On Sun, 30 Jan 2005, Remco Barende wrote:
You'll be better off buying the TDM400 from Digium.
I just ditched all my X100P clones because of huge problems with echoes.
Even worse, the interrupt problems with the clone cards caused my box to
crash under heavy I/O.
Want to buy the X100P
I have had problems with SCSI CD-ROMS. Is yours SCSI?
The best is an IDE CD-ROM on the primary IDE bus. Even
if you have a SCSI system you could get an IDE cdrom
for $10 just to load Asterisk.
-Andrew
--- Robert Augustyn [EMAIL PROTECTED] wrote:
Hi,
After I boot of the CD ( version 0.4 ) I
On Sun, 30 Jan 2005, Paul Tyreman wrote:
Hi,
This might not be a very popular question, but I was just wondering if
anyone have ever tried to run Asterisk on a Windows computer using Microsoft
Virtual Server
(http://www.microsoft.com/windowsserversystem/virtualserver/default.mspx).
AMP does not support ZAP entensions. only sip and iax.
Maybe in a future release. You could post to the AMP
site and ask for this feature.
For now you will need to hack extensions.conf to get
it to work.
--- Chuck Keeter [EMAIL PROTECTED] wrote:
At 07:57 PM 1/29/2005, you wrote:
;Extra FXS
You'll be better off buying the TDM400 from Digium.
I just ditched all my X100P clones because of huge problems with echoes.
Even worse, the interrupt problems with the clone cards caused my box to
crash under heavy I/O.
Want to buy the X100P clones I have? :)
About the best
Hi Ignacio,
Here the info: eyeBeam 1.1 3003x stamp 16296
Asterisk CVS-HEAD-12/05/04-09:27:10
Regards,
Francois
i did evrything you mentioned, i thing is for my eyebeam version, mine is
3002s
what`s yours?
On Fri, 28 Jan 2005 23:10:40 -0500 (EST), Francois Meehan
[EMAIL PROTECTED]
On 30 Jan 2005, at 14:02, Paul Tyreman wrote:
Hi,
This might not be a very popular question, but I was just wondering if
anyone have ever tried to run Asterisk on a Windows computer using
Microsoft Virtual Server
(http://www.microsoft.com/windowsserversystem/virtualserver/
default.mspx).
I
Another way to go would be to spend your upgrade money on one of these:
https://www.rpanetwork.co.uk/catalogue/basket.rhtm?
add=1productID=180544pi=1
It's a low powered server which runs asterisk quite nicely.
Installed one with an E1 card last week and it seems fine so far.
I had RPA
Want to buy the X100P clones I have? :)
About the best thing that you can do to reduce Echo on the X101P and X100P
cards is to;
1. Set the following in zapata.conf;
echocancel=128
echocancelwhenbridged=no
echotraining=800
2. Enable aggressive echo cancellation in zaptel's zconfig.h file;
/*
*
Hi,
I'd like to trigger call recording during call. Do I have any keys that can
be pressed during call ?
I've tried this, but doesn't start anything ( I guess that a is active
only during voicemail ?):
exten = a,1,DBget(temp=Record/${TIMESTAMP}_${UNIQUEID}_${CALLERID}) ;
Already recording ? if
Hi,
I made the mistake of ordering an 800 number as well. I have the same
problem you have, asterisk registers, but I get a fast busy when
dialing the number. Checking the cdr, and it shows the call has been
placed, but for 0.0 minutes.
I have yet to hear from them as well. Reading other posts,
On 30 Jan 2005, at 15:07, Michael 'Moose' Dinn wrote:
Another way to go would be to spend your upgrade money on one of
these:
https://www.rpanetwork.co.uk/catalogue/basket.rhtm?
add=1productID=180544pi=1
It's a low powered server which runs asterisk quite nicely.
Installed one with an E1 card
I am using AgentCallbacklogin to logon agents. I am
trying to avoid agents being logged in more than once in different extensions
(is this a bug?) by passing the callerid to the AgentCallbacklogin funtcion as
an option. The problem is thatby doing this, agents are not askedfor
an extension
iaxComm is an Open Source softphone for the Asterisk PBX.
iaxComm compiles and runs on Win32, Linux and Mac OS X (Panther) systems.
snip
http://iaxclient.sourceforge.net/iaxcomm-win-1.0rc1.zip
http://iaxclient.sourceforge.net/iaxcomm-lin-1.0rc1.tar
I've installed and used the new
Why does the X100P have echo and the Wildcard TDM400P have
no echo?
I thought the only advantage of using the TDM400P was that
it used less interrupts than the X100P?
Are there any other advantages?
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
exten = s,9,UserEvent(AgentMoreTime,Agent: ${agent}\r\nUntil: ${wrapupat});
Fragment \r\n parses into rn. \\r\\n turns into \r\n (uninterpreted).
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On January 30, 2005 09:31 am, Greg Boehnlein wrote:
Same reason people stick with Gentoo after a stage one installation. ;) I
have a theory about Gentoo that explains the rabid nature of Gentoo fans.
I believe that people that radically defend Gentoo and it's stage one
installation process are
http://www.digium.com/index.php?menu=astwind
I think this may be worth a look, I'm downloading it as I type this
e-mail...
I didn't know Asterisk had the possibility of being run on a windows machine
and while it's not as stable as a Linux implementation, it might just do for
the moment, as I
The digium X100P is probably fine. What most of these people
are talking about are the X100P clones which are of varying
qualities. I have had no problems, but results vary.
Cheers,
Jon.
On Sunday 30 January 2005 09:42 am, dean collins wrote:
Why does the X100P have echo and the Wildcard
Try the monitor application instead of record. I think that'll do what
you're looking for.
On Fri, 2005-01-28 at 13:30 -0800, David Shaw wrote:
Hello All,
I would like to record inbound and outbound calls to and from one
number.
I tried to add lines to my extensions.conf:
DAY=`date
Why does the X100P have echo and the Wildcard TDM400P have no echo?
I thought the only advantage of using the TDM400P was that it used
less interrupts than the X100P?
Are there any other advantages?
One of the major differences between the two cards is that tdm
fxo modules have support
LiveVoIP has many problems. #1 seems to be customer service. #2 they use
Level 3. I am switching to Txlink
On Sun, 2005-01-30 at 09:15, Ryan Laginski wrote:
Hi,
I made the mistake of ordering an 800 number as well. I have the same
problem you have, asterisk registers, but I get a fast busy
I have a separate extension set up for logoff that doesn't pass the
callerid to AgentCallbacklogin. So my agents dial 301 to log on and 302
to logoff. A lot of the phones (Cisco for example) allow you set up
speed dials, so its even easier for an agent to log on and off.
-Original
Hi,
I've found the Vertical Service Codes / vservices.inc of Julian in the cache
of google.
It's an very extended extensions include with all the *21 *67 etc services
implemented so it is stored to ODBC or if you replace it to Dbget/put etc.
I'm wondering if somebody has the macro/agi for using
When I place a call with asterisk, asterisk will try to dial
out on the first line even if the first line is already being
used by someone else. Any ideas on what I'm doing
wrong?
My question would be, how would asterisk know the line is in use if it
isn't controlling it?
Don't forget to warn your callers about the recording.
Tim Mattison wrote:
Try the monitor application instead of record. I think that'll do what
you're looking for.
On Fri, 2005-01-28 at 13:30 -0800, David Shaw wrote:
Hello All,
I would like to record inbound and outbound calls to and from one
Hi!,
I'm trying to set up a Queue (which works fine now :-)
Sip clients can login in to the Queue with dialing 91 on there phone.
And as soon as there are customers the Queue calls the agents back.
I would like that the queue calls the agents also if it's phone is
call-forwarded.
With agents
On Sat, 29 Jan 2005 21:17:56 -0500 (EST), Paul Dugas wrote:
This is probably going to sound really silly and I must be confused about
it. Maybe someone can set me straight.
I've been tinkering for a while with * and a number of different FXO/FXS
cards, SIP phones, and ATAs trying to get a feel
There is the little problem of having to switch numbers and then
communicating to everyone that the number has changed. This also only
seems to be a problem on inbound calls.
On Sat, 29 Jan 2005 23:34:49 -0500, Andrew Kohlsmith
[EMAIL PROTECTED] wrote:
On January 29, 2005 11:29 pm, Brian
Hello,
I'm attempting to setup a test asterisk server. I have a couple of old 4
port D/41D ISA dialogic cards. Has anyone had any success at getting them
to work? I've done some searching on the internet and it seems like some
have them working but I have not been able to find any help docs on
Thanks to everyone who responded with both advice and words of caution.
I have no doubt I am going to be getting in over my head. That is just my
M.O. but I'm still surviving an arguably better off for it.
The bottom line is this: my organization needs a phone system. We can't
afford
The best place for patches to Asterisk is the Asterisk bug tracker,
which can be found at http://bugs.digium.com .
*Please* don't forget to read the guidelines:
http://www.digium.com/bugguidelines.html and file a disclaimer.
Thanks for the patch! I'll look for it on the bug tracker :)
wow! excellent information, this should be added to the wiki under x100p /
tdm400.
Especially the info that the X100P is 600 ohm only would have made it a
lot clearer to me that I needn't buy an X100P (clone) for use in Europe
On Sun, 30 Jan 2005, Rich Adamson wrote:
Why does the X100P have
On Sun, 30 Jan 2005, Paul Tyreman wrote:
http://www.digium.com/index.php?menu=astwind
I think this may be worth a look, I'm downloading it as I type this
e-mail...
I didn't know Asterisk had the possibility of being run on a windows machine
and while it's not as stable as a Linux
Geoff Speicher wrote:
Sipura has implemented auto-answer in version 0.9.5 of the SPA-841
firmware. However, it is implemented via the Call-Info header, which
Asterisk stable doesn't currently support.
The attached patch implments a quick hack to support the Call-Info
header from the Dial()
Hi Greg,
On Sun, 2005-30-01 at 09:42 -0500, Greg Boehnlein wrote:
It works, but you will have timing issues and very poor audio quality.
I've run linux both under Vmware as well as running it under CoLinux
directly on windows w/ no emulation neccessary. All emulation /
virtualization
snip
Any work to support some USB Phones!? The ability to dial using the phones
keypad!?
Not yet, but I'll probably add suport for the TigerJet phone eventually.
/snip
That would be -awesome.- The mgmt at my company wants that ability, and
integrating with IAX protocol would make
We've got an office with a marginal broadband connection.
Do you think Asterisk + Grandstreams can be tuned to give a good quality
call on with this jitter/latency ? This is the only broadband supplier
in the area, so changing carriers would be difficult.
I made a call from the Grandstream with
Were the pops/drops/buzzes a problem only with communications via a
telephony card?
We ran Asterisk in a VPC instance (Redhat 8.0) for 3 months while we
evaluated Asterisk. The only reason we had to move to a version of Linux
running directly on hardware was a need to run X101P cards.
We had
On January 30, 2005 12:18 pm, Brian Dingman wrote:
There is the little problem of having to switch numbers and then
communicating to everyone that the number has changed. This also only
seems to be a problem on inbound calls.
And why, praytell, did you go into production with DIDs from a
Hello everybody!
I am having the following problem and since I am a beginner in playing
with asterisk, i can't solve it:
I am trying to integrate my existing H.323 network in real world
telephony by ISDN cards. The problem is that i DON'T want to change all
e164 numbers in my h.323 network and
John Middleton wrote:
Anyone got any experiences of these with *, and also costings?
Someone mentioned them on the list several months ago, but I don't think
anyone mentioned actually using it.
--
Andrew Thompson
http://aktzero.com/
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Has anyone benchmarked Asterisk on a dedicated single versus dual
processor machine? Or could any Asterisk developers comment on whether
it is architected in such a way that threads could run on multiple CPUs
(especially MeetMe2)?
At a higher level, can I host more simultaneous lines and/or
Can't asterisk look for a dialtone? Even a $5 modem
can detect whether or not there is a dialtone.
Thanks,
Jon.
On Sunday 30 January 2005 10:37 am, Wilson Pickett wrote:
When I place a call with asterisk, asterisk will try to dial
out on the first line even if the first line is already
Calin Serbanescu schrieb:
...
e164 numbers in my h.323 network and my ISDN provider doesn't accept
those identities (CIDs). So, i have to spoof the outgoing CID depending
on incoming CID.
Is there any possible way of doing this by AGI? How? any examples are
welcome.
Hi,
I'm not sure, whether
unfortunatelly i have to accept their terms and rewrite caller id... but
again, i am newbie in scripting with agi and i can't find any example on
the net about this... do you have any link to such script ?
thanks
On Sun, 2005-01-30 at 20:42 +0100, Roger Schreiter wrote:
Calin Serbanescu
i have the same problem...
i've also added a feature request to bug tracker
(http://bugs.digium.com/bug_view_page.php?bug_id=0002612) regarding
this issue.
On Sun, 30 Jan 2005 13:40:06 -0600, Jon Gabrielson
[EMAIL PROTECTED] wrote:
Can't asterisk look for a dialtone? Even a $5 modem
can
Calin Serbanescu schrieb:
...
the net about this... do you have any link to such script ?
No, I don't have such a link.
But on the voip-wiki pages there are some examples
for agi-scipting.
There are APIs for some common languages, e.g. Perl,
which is maybe one of the fastet ways to code simple
thanks very much, i'll try that... that is, no sleep until it's ready :)
On Sun, 2005-01-30 at 21:17 +0100, Roger Schreiter wrote:
Calin Serbanescu schrieb:
...
the net about this... do you have any link to such script ?
No, I don't have such a link.
But on the voip-wiki pages there
Calin Serbanescu schrieb:
unfortunatelly i have to accept their terms and rewrite caller id... but
again, i am newbie in scripting with agi and i can't find any example on
the net about this... do you have any link to such script ?
... what I should maybe also mention:
My script in the recent
Hi there,
this is just a short note about one of the PA168x based phones out there
which I obtained as Giptel G100 (aka Siptronic ST-100): For some reason
this phone would refuse to register with Asterisk using SIP, but after
uploading the IAX2 firmware instead it finally came to life:
Hi list!
I'm still trying to figure out about the groups in asterisk.
If I understand correctly, if you assign a certain group number and you
assign the same call group number to a sip device the device will reing
even though you did not specifically specify it in extension.conf?
How will this
Hi!
wow! excellent information, this should be added to the wiki under
x100p / tdm400.
So did _you_ add it?
Cheers, Philipp
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To
Dhennys ,
Estou interessado no SIPURA também. Você já tem alguma idéia em como adquirir?
Estou em São Carlos, SP. Podemos, eventualmente, dividir custos de envio.
--
Antonio José dos Santos Brandão
On Sat, 29 Jan 2005 05:20:30 -0200, Dhennys Pestana [EMAIL PROTECTED] wrote:
(I appologize
I spent several days last year trying to get a TE410 to work on an HP DL360,
but never seemed to get it to work.
I called Digium's (usually great) tech support at the time, and their response
to me was 'if you can't get it to work we can't help you'.
Please let me know if you have any
Hi Friends
I have a problem presenting Caller ID on my H323 GW.
Scenario: Sip
Phone Asterisk H323 GW PSTN (E1)
From PSTN to the Sip phone works fine I put this lines in extentions.conf
exten = 1234,1,SetCallerID,
${CALLERIDNUM}
exten = 1234,2,Dial(${testphone1},20,Ttm)
I can send calls from asterisk to a Sipura FXO
interface (SIP/300) from any SIP phones including SIP/205 which is the Sipura
3000 FXS interface.
The problem I have is when a call from the PSTN is sends to Asterisk. On
extnesion conf I dial all the SIP clientsI get a 302 Moved temporarily
Depending on your state. :P
On Sun, 2005-01-30 at 11:42 -0500, Mark Phillips wrote:
Don't forget to warn your callers about the recording.
Tim Mattison wrote:
Try the monitor application instead of record. I think that'll do what
you're looking for.
On Fri, 2005-01-28 at 13:30
On Sun, 2005-01-30 at 13:40 -0600, Jon Gabrielson wrote:
Can't asterisk look for a dialtone? Even a $5 modem
can detect whether or not there is a dialtone.
Maybe you should just use your $5 modem and write your own software.
Asterisk is a PBX. PBXs shouldn't have to deal with your
or maybe country? or should that be County? :)
Mike
On Sun, 30 Jan 2005 16:49:29 -0500, Tim Mattison [EMAIL PROTECTED] wrote:
Depending on your state. :P
On Sun, 2005-01-30 at 11:42 -0500, Mark Phillips wrote:
Don't forget to warn your callers about the recording.
Tim Mattison
hi;
have couple of questions regarding meet_me conference room application:
1) is there a maximum allowable number of concurrently active conference
rooms per server?
2) what is the maximum allowable number of users in a given conference
room before quailty creeps out?
thanks
moe smadi
Good call.
For our American readers... does anyone know where I can obtain a list
of states/counties and their regulations in regards to call recording?
On Sun, 2005-01-30 at 22:10 +, Mike Dent wrote:
or maybe country? or should that be County? :)
Mike
On Sun, 30 Jan 2005 16:49:29
I get these errors, and i am stuck here. I don't know what to do.
I am using latest stable Zaptel 1.0.4 and Asterisk 1.0.5. I just ran it on a
freinds machine and it went well there.
chan_zap.c:3647: dereferencing pointer to incomplete type
chan_zap.c:3648: dereferencing pointer to incomplete
Hi all,
We're in a transition between OldPhoneSystem and Asterisk. One of the
things that's needed to be done right now with OldPhoneSystem is the
ability to record calls. I thought Asterisk can record calls, so I
set about to make it happen. And it does, sort of.
I made a .call file that rings
On Sunday 30 January 2005 23:29, [EMAIL PROTECTED] wrote:
I get these errors, and i am stuck here. I don't know what to do.
I am using latest stable Zaptel 1.0.4 and Asterisk 1.0.5. I just ran it on
a freinds machine and it went well there.
[...]
chan_zap.c:3669: dereferencing pointer to
I am getting the following error when compiling oh323-0.7.1 with
Asterisk CVS (2004-12-21: Updated versions 0.7.1 (for Asterisk CVS
HEAD)
make[1]: Leaving directory `/usr/src/asterisk-oh323-0.7.1/wrapper'
make: *** [subdirs_build] Error 1
bash-2.05b# asteriskaudio.cxx:167: (Each undeclared
Asterisk should be able to do this, there are several cases
when this is essential. The first is a shared/party line where
asterisk cannot have guaranteed access for whatever reason.
In our case, that reason happens to be because we also use
our outgoing lines for faxing.
The second is that
Duane wrote:
Adam Hart wrote:
As always, I'm happy to announce a new version of Firefly.
Firefly 1.9.8 has more of what you want and less of what you don't
http://www.virbiage.com/firefly/download/firefly-thirdparty.exe
There's a few bug fixes - notably fixed the Reject button and sending
of
Adam Hart wrote:
Few people have claimed success, I'm not sure how though.
Any chance of a native linux version then? :)
--
Best regards,
Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com -
jurgen wrote:
snip
Problem is, Asterisk times out and disconnects after 10 seconds,
stopping the recording.
If I run something else in the context, say the infamous Monkey
Sounds, everything's fine, and the call just keeps going, annoying the
people on the line with monkey sounds. For some reason,
I have updated the Wiki with this info as I have seen it come up a few
times.
Nathan.
Gary wrote:
Don't forget Howard, that Caller-ID presentation is an extra chargeable
service.
has it been turned on on these lines and confirmed ??
(its handy to carry a caller-id in your kit for checking:-)
On
So I have a problem. A customer of mine wants a PBX, owns an
office building. I want to sell him on asterisk. He has 4 tenants. I am
using my asterisk box to simulate it. My asterisk box has a TDM400P card, not a
PRI card. Dont know if it makes any difference.
Anyway, I want to route
Hi,
I've installed meetme2 according to instructions. Everything seems ok,
members of conference are displayed, but nothing happens if I click on 3
action buttons (kick out, talklisten, ...).
Any hints how to deal with this ? In what way exactly does meetme2 kick
user off the conference ?
Jason Brown wrote:
Now I understand it is looking for the startup point. I dont understand
why. 2 other asterisk guys I know swear its supposed to work, although
they are using sip/iax and not zap for input.
And why would you think those would act similarly? They don't.
Zap channels without ISDN
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